[pjsip] Program crash when hungup

Somnuk Wongkhan wearetherock at gmail.com
Wed Aug 19 01:03:42 EDT 2009


I also attach source code (not difference from
http://www.pjsip.org/pjsip/docs/html/page_pjsip_sample_simple_pjsuaua_c.htm)

When start program with "myapp.exe
sip:2222 at 192.168.1.106<sip%3A2222 at 192.168.1.106>"
it work well
this program make call to another phone.  hangup from 2222 is ok

But make call from 2222 to this program then hang it up , it crash (program
terminate)
I don't know what wrong because i copy this code from
http://www.pjsip.org/pjsip/docs/html/page_pjsip_sample_simple_pjsuaua_c.htm

I compile myapp.cpp with MinGW under Windows 7.
Anyone can help? Thank.

Last message before crash
--end msg--
 11:38:06.769            APP  Call 1 state=CONFIRMED
 11:38:06.809    ec0x1a9d688  Underflow, buf_cnt=0, will generate 1 frame
 11:38:07.289  strm0x1a991b4  VAD re-enabled
 11:38:28.819   pjsua_core.c  RX 535 bytes Request msg OPTIONS/cseq=102
(rdata0x1a88864) from UDP 192.168.1.106:5060:
OPTIONS sip:1111 at 192.168.1.100:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK2b9f16c4;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown at 192.168.1.106 <sip%3AUnknown at 192.168.1.106>
>;tag=as577732a7
To: <sip:1111 at 192.168.1.100:5060>
Contact: <sip:Unknown at 192.168.1.106 <sip%3AUnknown at 192.168.1.106>>
Call-ID: 26ae8aee33ec93a37ca4095a663299d0 at 192.168.1.106
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.9-samy-r27
Date: Wed, 19 Aug 2009 04:38:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


--end msg--
 11:38:34.053   Master/sound  Underflow, buf_cnt=0, will generate 1 frame
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/*
#include <pjlib.h>
#include <pjlib-util.h>
#include <pjmedia.h>
#include <pjmedia-codec.h>
#include <pjsip.h>
#include <pjsip_simple.h>
#include <pjsip_ua.h>
#include <pjsua-lib/pjsua.h>
*/

#include <pjsua-lib/pjsua.h>
#define THIS_FILE "APP"
#define SIP_DOMAIN	"192.168.1.106"
#define SIP_USER	"1111"
#define SIP_PASSWD	"1111"

static void on_incoming_call(pjsua_acc_id acc_id, pjsua_call_id call_id, pjsip_rx_data *rdata){
	
	pjsua_call_info ci;
	PJ_UNUSED_ARG(acc_id);
	PJ_UNUSED_ARG(rdata);
	
	pjsua_call_get_info(call_id, &ci);
	PJ_LOG(3,(THIS_FILE, "Incomming call from %.*s!!",
		(int)ci.remote_info.slen, ci.remote_info.ptr));
	
	pjsua_call_answer(call_id, 200, NULL, NULL);
}
static void on_call_state(pjsua_call_id call_id, pjsip_event *e){
	pjsua_call_info ci;
	PJ_UNUSED_ARG(e);
	pjsua_call_get_info(call_id, &ci);
	PJ_LOG(3,(THIS_FILE, "Call %d state=%.*s", call_id,
		(int) ci.state_text.slen, ci.state_text.ptr));
}
static void on_call_media_state(pjsua_call_id call_id){
	pjsua_call_info ci;
	pjsua_call_get_info(call_id, &ci);
	if (ci.media_status == PJSUA_CALL_MEDIA_ACTIVE) {
		pjsua_conf_connect(ci.conf_slot, 0);
		pjsua_conf_connect(0, ci.conf_slot);
	}
}
static void error_exit(const char *title, pj_status_t status){
	pjsua_perror(THIS_FILE, title, status);
	pjsua_destroy();
	exit(1);
}
int main(int argc, char *argv[])
{
	pjsua_acc_id acc_id;
	pj_status_t status;
	
	status = pjsua_create();
	if(status != PJ_SUCCESS) error_exit("Error in pjsua_create()", status);
	
	if(argc >1){
		status = pjsua_verify_sip_url(argv[1]);
		if(status != PJ_SUCCESS) error_exit("Invalid URL in argv", status);
	}
	
	{
		pjsua_config cfg;
		pjsua_logging_config log_cfg;
		pjsua_config_default(&cfg);
		cfg.cb.on_incoming_call = &on_incoming_call;
		cfg.cb.on_call_media_state = &on_call_media_state;
		cfg.cb.on_call_state = &on_call_state;
		
		pjsua_logging_config_default(&log_cfg);
		log_cfg.console_level = 4;
		
		status = pjsua_init(&cfg, &log_cfg, NULL);
		if (status != PJ_SUCCESS) error_exit("Error in pjsua_init()", status);
	}
	
	/* Add UDP transport.*/
	{
		pjsua_transport_config cfg;
		pjsua_transport_config_default(&cfg);
		cfg.port = 5060;
		status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &cfg, NULL);
		if(status != PJ_SUCCESS) error_exit("Error creating transport", status);
	}
	
	/* Initialization is done, now start pjsua */
	status = pjsua_start() ;
	if(status != PJ_SUCCESS) error_exit("Error start pjsua", status);
	
	/* Register to SIP server by creating SIP account. */
	{
		pjsua_acc_config cfg;
		pjsua_acc_config_default(&cfg);
		cfg.id = pj_str("sip:" SIP_USER "@" SIP_DOMAIN);
		cfg.reg_uri = pj_str("sip:" SIP_DOMAIN);
		cfg.cred_count = 1 ;
		cfg.cred_info[0].realm = pj_str("*");
		cfg.cred_info[0].scheme = pj_str("digest");
		cfg.cred_info[0].username = pj_str(SIP_USER);
		cfg.cred_info[0].data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;
		cfg.cred_info[0].data = pj_str(SIP_PASSWD);
		
		status = pjsua_acc_add(&cfg, PJ_TRUE, &acc_id);
		if(status != PJ_SUCCESS) error_exit("Error adding account", status);
	}
	
	/* If URI is specified , make call to the URL. */
	if(argc > 1){
		pj_str_t uri = pj_str(argv[1]);
		status = pjsua_call_make_call(acc_id, &uri, 0, NULL, NULL, NULL);
		if(status != PJ_SUCCESS) error_exit("Error making call", status);
	}
	
	/* Wait util user press "q" to quit. */
	for(;;){
		char option[10];
		puts("Press 'h' to hangup all calls, 'q' to quit");
		if(fgets(option, sizeof(option), stdin) == NULL){
			puts("EOF while reading stdin, will quit now..");
			break;
		}
		if(option[0] == 'q')
			break;
		if(option[0] == 'h')
			pjsua_call_hangup_all();
	}
	pjsua_destroy();
	return 0;
}
-------------- next part --------------
 11:42:27.851 os_core_win32.  pjlib 1.3 for win32 initialized
 11:42:27.875 sip_endpoint.c  Creating endpoint instance...
 11:42:27.875          pjlib  select() I/O Queue created (0x39bca0)
 11:42:27.875 sip_endpoint.c  Module "mod-msg-print" registered
 11:42:27.875 sip_transport.  Transport manager created.
 11:42:27.875 sip_endpoint.c  Module "mod-pjsua-log" registered
 11:42:27.875 sip_endpoint.c  Module "mod-tsx-layer" registered
 11:42:27.875 sip_endpoint.c  Module "mod-stateful-util" registered
 11:42:27.875 sip_endpoint.c  Module "mod-ua" registered
 11:42:27.875 sip_endpoint.c  Module "mod-100rel" registered
 11:42:27.876 sip_endpoint.c  Module "mod-pjsua" registered
 11:42:27.876 sip_endpoint.c  Module "mod-invite" registered
 11:42:27.917       pa_dev.c  PortAudio sound library initialized, status=0
 11:42:27.917       pa_dev.c  PortAudio host api count=2
 11:42:27.917       pa_dev.c  Sound device count=5
 11:42:27.923     wmme_dev.c  WMME initialized, found 4 devices:
 11:42:27.923     wmme_dev.c   dev_id 0: Wave mapper  (in=2, out=2)
 11:42:27.923     wmme_dev.c   dev_id 1: Microphone (High Definition Aud  (in=2, out=0)
 11:42:27.923     wmme_dev.c   dev_id 2: Line In (High Definition Audio   (in=2, out=0)
 11:42:27.923     wmme_dev.c   dev_id 3: Speakers (High Definition Audio  (in=0, out=2)
 11:42:27.924          pjlib  select() I/O Queue created (0x1b59d6c)
 11:42:27.924        libsrtp  Ugh: /dev/urandom not present, using rand() instead
 11:42:27.926        libsrtp  Ugh: /dev/urandom not present, using rand() instead
 11:42:27.927        libsrtp  Ugh: /dev/urandom not present, using rand() instead
 11:42:27.928        libsrtp  Ugh: /dev/urandom not present, using rand() instead
 11:42:27.929 sip_endpoint.c  Module "mod-evsub" registered
 11:42:27.929 sip_endpoint.c  Module "mod-presence" registered
 11:42:27.929 sip_endpoint.c  Module "mod-refer" registered
 11:42:27.929 sip_endpoint.c  Module "mod-pjsua-pres" registered
 11:42:27.929 sip_endpoint.c  Module "mod-pjsua-im" registered
 11:42:27.929 sip_endpoint.c  Module "mod-pjsua-options" registered
 11:42:27.930   pjsua_core.c  1 SIP worker threads created
 11:42:27.930   pjsua_core.c  pjsua version 1.3 for i686-pc-mingw32 initialized
 11:42:27.931   pjsua_core.c  SIP UDP socket reachable at 192.168.1.100:5060
 11:42:27.931   udp0x1b5f0a8  SIP UDP transport started, published address is 192.168.1.100:5060
 11:42:27.932  pjsua_media.c  RTP socket reachable at 192.168.1.100:4000
 11:42:27.932  pjsua_media.c  RTCP socket reachable at 192.168.1.100:4001
 11:42:27.933  pjsua_media.c  RTP socket reachable at 192.168.1.100:4002
 11:42:27.933  pjsua_media.c  RTCP socket reachable at 192.168.1.100:4003
 11:42:27.934  pjsua_media.c  RTP socket reachable at 192.168.1.100:4004
 11:42:27.934  pjsua_media.c  RTCP socket reachable at 192.168.1.100:4005
 11:42:27.937  pjsua_media.c  RTP socket reachable at 192.168.1.100:4006
 11:42:27.937  pjsua_media.c  RTCP socket reachable at 192.168.1.100:4007
 11:42:27.937    pjsua_acc.c  Account sip:1111 at 192.168.1.106 added with id 0
 11:42:27.938   pjsua_core.c  TX 386 bytes Request msg REGISTER/cseq=34513 (tdta0x1b77838) to UDP 192.168.1.106:5060:
REGISTER sip:192.168.1.106 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.100:5060;rport;branch=z9hG4bKPjcdb582acdca44a1290bf39935c2a2d3f

Max-Forwards: 70

From: <sip:1111 at 192.168.1.106>;tag=94ee42002adc4cf2894281cb11da935c

To: <sip:1111 at 192.168.1.106>

Call-ID: 02e3fa1a88cf4317a99db0d69b5a48e9

CSeq: 34513 REGISTER

Contact: <sip:1111 at 192.168.1.100:5060>

Expires: 300

Content-Length:  0




--end msg--
 11:42:27.938    pjsua_acc.c  Registration sent
 11:42:27.938  pjsua_media.c  Opening sound device PCM at 16000/1/20ms
 11:42:27.943   pjsua_core.c  RX 561 bytes Response msg 401/REGISTER/cseq=34513 (rdata0x1b68864) from UDP 192.168.1.106:5060:
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bKPjcdb582acdca44a1290bf39935c2a2d3f;received=192.168.1.100;rport=5060

From: <sip:1111 at 192.168.1.106>;tag=94ee42002adc4cf2894281cb11da935c

To: <sip:1111 at 192.168.1.106>;tag=as6d46783e

Call-ID: 02e3fa1a88cf4317a99db0d69b5a48e9

CSeq: 34513 REGISTER

User-Agent: Asterisk PBX 1.6.0.9-samy-r27

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="33aa239f"

Content-Length: 0




--end msg--
 11:42:27.944   pjsua_core.c  TX 546 bytes Request msg REGISTER/cseq=34514 (tdta0x1b77838) to UDP 192.168.1.106:5060:
REGISTER sip:192.168.1.106 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.100:5060;rport;branch=z9hG4bKPjbc6c1075678a468191ad325e6d9ae705

Max-Forwards: 70

From: <sip:1111 at 192.168.1.106>;tag=94ee42002adc4cf2894281cb11da935c

To: <sip:1111 at 192.168.1.106>

Call-ID: 02e3fa1a88cf4317a99db0d69b5a48e9

CSeq: 34514 REGISTER

Contact: <sip:1111 at 192.168.1.100:5060>

Expires: 300

Authorization: Digest username="1111", realm="asterisk", nonce="33aa239f", uri="sip:192.168.1.106", response="1722c88c9aa369ec0fc019b0639face8", algorithm=MD5

Content-Length:  0




--end msg--
 11:42:27.947   pjsua_core.c  RX 535 bytes Request msg OPTIONS/cseq=102 (rdata0x1b68864) from UDP 192.168.1.106:5060:
OPTIONS sip:1111 at 192.168.1.100:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK0452b14f;rport

Max-Forwards: 70

From: "Unknown" <sip:Unknown at 192.168.1.106>;tag=as6611d81e

To: <sip:1111 at 192.168.1.100:5060>

Contact: <sip:Unknown at 192.168.1.106>

Call-ID: 1c1f29b871e9de0a1353986b74b4a081 at 192.168.1.106

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX 1.6.0.9-samy-r27

Date: Wed, 19 Aug 2009 04:42:52 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces, timer

Content-Length: 0




--end msg--
 11:42:28.008    ec0x1b7dcb0  AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=200 ms, latency=100 ms
 11:42:28.008   pjsua_call.c  Making call with acc #0 to sip:2222 at 192.168.1.106
 11:42:28.009  pjsua_media.c  Media index 0 selected for call 0
 11:42:28.009   pjsua_core.c  TX 998 bytes Request msg INVITE/cseq=17311 (tdta0x1bade88) to UDP 192.168.1.106:5060:
INVITE sip:2222 at 192.168.1.106 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.100:5060;rport;branch=z9hG4bKPjb3b0682c5e5d4575b603d34f615f2300

Max-Forwards: 70

From: sip:1111 at 192.168.1.106;tag=27b9f31d7d084a55ac6daff93aa6f342

To: sip:2222 at 192.168.1.106

Contact: <sip:1111 at 192.168.1.100:5060>

Call-ID: 74fac184beef419eb84af5e932b8a8e5

CSeq: 17311 INVITE

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

Supported: replaces, 100rel, norefersub

Content-Type: application/sdp

Content-Length:   462



v=0

o=- 3459670948 3459670948 IN IP4 192.168.1.100

s=pjmedia

c=IN IP4 192.168.1.100

t=0 0

a=X-nat:0

m=audio 4000 RTP/AVP 103 102 104 113 3 0 8 9 101

a=rtcp:4001 IN IP4 192.168.1.100

a=rtpmap:103 speex/16000

a=rtpmap:102 speex/8000

a=rtpmap:104 speex/32000

a=rtpmap:113 iLBC/8000

a=fmtp:113 mode=30

a=rtpmap:3 GSM/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:9 G722/8000

a=sendrecv

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15


--end msg--
 11:42:28.009            APP  Call 0 state=CALLING
Press 'h' to hangup all calls, 'q' to quit
 11:42:28.009   pjsua_core.c  TX 1090 bytes Response msg 200/OPTIONS/cseq=102 (tdta0x1bb0640) to UDP 192.168.1.106:5060:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.106:5060;rport=5060;received=192.168.1.106;branch=z9hG4bK0452b14f

Call-ID: 1c1f29b871e9de0a1353986b74b4a081 at 192.168.1.106

From: "Unknown" <sip:Unknown at 192.168.1.106>;tag=as6611d81e

To: <sip:1111 at 192.168.1.100>

CSeq: 102 OPTIONS

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

Accept: application/sdp, application/pidf+xml, application/xpidf+xml, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain

Supported: replaces, 100rel, norefersub

Allow-Events: presence, refer

Content-Type: application/sdp

Content-Length:   451



v=0

o=- 3459670948 3459670948 IN IP4 192.168.1.100

s=pjmedia

c=IN IP4 192.168.1.100

t=0 0

m=audio 4000 RTP/AVP 103 102 104 113 3 0 8 9 101

a=rtcp:4001 IN IP4 192.168.1.100

a=rtpmap:103 speex/16000

a=rtpmap:102 speex/8000

a=rtpmap:104 speex/32000

a=rtpmap:113 iLBC/8000

a=fmtp:113 mode=30

a=rtpmap:3 GSM/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:9 G722/8000

a=sendrecv

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15


--end msg--
 11:42:28.010   pjsua_core.c  RX 578 bytes Response msg 200/REGISTER/cseq=34514 (rdata0x1b68864) from UDP 192.168.1.106:5060:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bKPjbc6c1075678a468191ad325e6d9ae705;received=192.168.1.100;rport=5060

From: <sip:1111 at 192.168.1.106>;tag=94ee42002adc4cf2894281cb11da935c

To: <sip:1111 at 192.168.1.106>;tag=as6d46783e

Call-ID: 02e3fa1a88cf4317a99db0d69b5a48e9

CSeq: 34514 REGISTER

User-Agent: Asterisk PBX 1.6.0.9-samy-r27

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces, timer

Expires: 300

Contact: <sip:1111 at 192.168.1.100:5060>;expires=300

Date: Wed, 19 Aug 2009 04:42:52 GMT

Content-Length: 0




--end msg--
 11:42:28.010    pjsua_acc.c  sip:1111 at 192.168.1.106: registration success, status=200 (OK), will re-register in 300 seconds
 11:42:28.010    pjsua_acc.c  Keep-alive timer started for acc 0, destination:192.168.1.106:5060, interval:15s
 11:42:28.012   pjsua_core.c  RX 555 bytes Response msg 401/INVITE/cseq=17311 (rdata0x1b68864) from UDP 192.168.1.106:5060:
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bKPjb3b0682c5e5d4575b603d34f615f2300;received=192.168.1.100;rport=5060

From: sip:1111 at 192.168.1.106;tag=27b9f31d7d084a55ac6daff93aa6f342

To: sip:2222 at 192.168.1.106;tag=as779078f4

Call-ID: 74fac184beef419eb84af5e932b8a8e5

CSeq: 17311 INVITE

User-Agent: Asterisk PBX 1.6.0.9-samy-r27

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="23761d3e"

Content-Length: 0




--end msg--
 11:42:28.012   pjsua_core.c  TX 338 bytes Request msg ACK/cseq=17311 (tdta0x1bb0640) to UDP 192.168.1.106:5060:
ACK sip:2222 at 192.168.1.106 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.100:5060;rport;branch=z9hG4bKPjb3b0682c5e5d4575b603d34f615f2300

Max-Forwards: 70

From: sip:1111 at 192.168.1.106;tag=27b9f31d7d084a55ac6daff93aa6f342

To: sip:2222 at 192.168.1.106;tag=as779078f4

Call-ID: 74fac184beef419eb84af5e932b8a8e5

CSeq: 17311 ACK

Content-Length:  0




--end msg--
 11:42:28.012   pjsua_core.c  TX 1163 bytes Request msg INVITE/cseq=17312 (tdta0x1bade88) to UDP 192.168.1.106:5060:
INVITE sip:2222 at 192.168.1.106 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.100:5060;rport;branch=z9hG4bKPj7f1601f6f7b242b5a1067a1a149ed32c

Max-Forwards: 70

From: sip:1111 at 192.168.1.106;tag=27b9f31d7d084a55ac6daff93aa6f342

To: sip:2222 at 192.168.1.106

Contact: <sip:1111 at 192.168.1.100:5060>

Call-ID: 74fac184beef419eb84af5e932b8a8e5

CSeq: 17312 INVITE

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

Supported: replaces, 100rel, norefersub

Authorization: Digest username="1111", realm="asterisk", nonce="23761d3e", uri="sip:2222 at 192.168.1.106", response="b504983e43d6d256f3e46f1b2744c28a", algorithm=MD5

Content-Type: application/sdp

Content-Length:   462



v=0

o=- 3459670948 3459670948 IN IP4 192.168.1.100

s=pjmedia

c=IN IP4 192.168.1.100

t=0 0

a=X-nat:0

m=audio 4000 RTP/AVP 103 102 104 113 3 0 8 9 101

a=rtcp:4001 IN IP4 192.168.1.100

a=rtpmap:103 speex/16000

a=rtpmap:102 speex/8000

a=rtpmap:104 speex/32000

a=rtpmap:113 iLBC/8000

a=fmtp:113 mode=30

a=rtpmap:3 GSM/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:9 G722/8000

a=sendrecv

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15


--end msg--
 11:42:28.013            APP  Call 0 state=CALLING
 11:42:28.018   pjsua_core.c  RX 493 bytes Response msg 100/INVITE/cseq=17312 (rdata0x1b68864) from UDP 192.168.1.106:5060:
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bKPj7f1601f6f7b242b5a1067a1a149ed32c;received=192.168.1.100;rport=5060

From: sip:1111 at 192.168.1.106;tag=27b9f31d7d084a55ac6daff93aa6f342

To: sip:2222 at 192.168.1.106

Call-ID: 74fac184beef419eb84af5e932b8a8e5

CSeq: 17312 INVITE

User-Agent: Asterisk PBX 1.6.0.9-samy-r27

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces, timer

Contact: <sip:2222 at 192.168.1.106>

Content-Length: 0




--end msg--
 11:42:28.539   pjsua_core.c  RX 509 bytes Response msg 180/INVITE/cseq=17312 (rdata0x1b68864) from UDP 192.168.1.106:5060:
SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bKPj7f1601f6f7b242b5a1067a1a149ed32c;received=192.168.1.100;rport=5060

From: sip:1111 at 192.168.1.106;tag=27b9f31d7d084a55ac6daff93aa6f342

To: sip:2222 at 192.168.1.106;tag=as7b91226b

Call-ID: 74fac184beef419eb84af5e932b8a8e5

CSeq: 17312 INVITE

User-Agent: Asterisk PBX 1.6.0.9-samy-r27

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces, timer

Contact: <sip:2222 at 192.168.1.106>

Content-Length: 0




--end msg--
 11:42:28.539            APP  Call 0 state=EARLY
 11:42:28.578   pjsua_core.c  RX 509 bytes Response msg 180/INVITE/cseq=17312 (rdata0x1b68864) from UDP 192.168.1.106:5060:
SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bKPj7f1601f6f7b242b5a1067a1a149ed32c;received=192.168.1.100;rport=5060

From: sip:1111 at 192.168.1.106;tag=27b9f31d7d084a55ac6daff93aa6f342

To: sip:2222 at 192.168.1.106;tag=as7b91226b

Call-ID: 74fac184beef419eb84af5e932b8a8e5

CSeq: 17312 INVITE

User-Agent: Asterisk PBX 1.6.0.9-samy-r27

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces, timer

Contact: <sip:2222 at 192.168.1.106>

Content-Length: 0




--end msg--
 11:42:28.578            APP  Call 0 state=EARLY
 11:42:32.481   pjsua_core.c  RX 835 bytes Response msg 200/INVITE/cseq=17312 (rdata0x1b68864) from UDP 192.168.1.106:5060:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bKPj7f1601f6f7b242b5a1067a1a149ed32c;received=192.168.1.100;rport=5060

From: sip:1111 at 192.168.1.106;tag=27b9f31d7d084a55ac6daff93aa6f342

To: sip:2222 at 192.168.1.106;tag=as7b91226b

Call-ID: 74fac184beef419eb84af5e932b8a8e5

CSeq: 17312 INVITE

User-Agent: Asterisk PBX 1.6.0.9-samy-r27

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces, timer

Contact: <sip:2222 at 192.168.1.106>

Content-Type: application/sdp

Content-Length: 298



v=0

o=root 1075496772 1075496772 IN IP4 192.168.1.106

s=Asterisk PBX 1.6.0.9-samy-r27

c=IN IP4 192.168.1.106

t=0 0

m=audio 11456 RTP/AVP 0 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv


--end msg--
 11:42:32.481            APP  Call 0 state=CONNECTING
 11:42:32.482  strm0x1bb2f64  VAD temporarily disabled
 11:42:32.482  strm0x1bb2f64  Encoder stream started
 11:42:32.482  strm0x1bb2f64  Decoder stream started
 11:42:32.482  pjsua_media.c  Media updates, stream #0: PCMU (sendrecv)
 11:42:32.483   conference.c  Port 1 (sip:2222 at 192.168.1.106) transmitting to port 0 (Microsoft Sound Mapper - Input)
 11:42:32.483   conference.c  Port 0 (Microsoft Sound Mapper - Input) transmitting to port 1 (sip:2222 at 192.168.1.106)
 11:42:32.483   pjsua_core.c  TX 338 bytes Request msg ACK/cseq=17312 (tdta0x1bb6610) to UDP 192.168.1.106:5060:
ACK sip:2222 at 192.168.1.106 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.100:5060;rport;branch=z9hG4bKPjf34ed215c1cc43d3b235d7f73b136980

Max-Forwards: 70

From: sip:1111 at 192.168.1.106;tag=27b9f31d7d084a55ac6daff93aa6f342

To: sip:2222 at 192.168.1.106;tag=as7b91226b

Call-ID: 74fac184beef419eb84af5e932b8a8e5

CSeq: 17312 ACK

Content-Length:  0




--end msg--
 11:42:32.483            APP  Call 0 state=CONFIRMED
 11:42:32.485   Master/sound  Underflow, buf_cnt=0, will generate 1 frame
 11:42:33.116  strm0x1bb2f64  VAD re-enabled
 11:42:40.434    ec0x1b7dcb0  Underflow, buf_cnt=0, will generate 1 frame
 11:42:43.352   pjsua_core.c  RX 430 bytes Request msg BYE/cseq=102 (rdata0x1b68864) from UDP 192.168.1.106:5060:
BYE sip:1111 at 192.168.1.100:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK591a44dd;rport

Max-Forwards: 70

From: sip:2222 at 192.168.1.106;tag=as7b91226b

To: sip:1111 at 192.168.1.106;tag=27b9f31d7d084a55ac6daff93aa6f342

Call-ID: 74fac184beef419eb84af5e932b8a8e5

CSeq: 102 BYE

User-Agent: Asterisk PBX 1.6.0.9-samy-r27

X-Asterisk-HangupCause: Normal Clearing

X-Asterisk-HangupCauseCode: 16

Content-Length: 0




--end msg--
 11:42:43.352   pjsua_core.c  TX 304 bytes Response msg 200/BYE/cseq=102 (tdta0x1b77838) to UDP 192.168.1.106:5060:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.106:5060;rport=5060;received=192.168.1.106;branch=z9hG4bK591a44dd

Call-ID: 74fac184beef419eb84af5e932b8a8e5

From: <sip:2222 at 192.168.1.106>;tag=as7b91226b

To: <sip:1111 at 192.168.1.106>;tag=27b9f31d7d084a55ac6daff93aa6f342

CSeq: 102 BYE

Content-Length:  0




--end msg--
 11:42:43.353            APP  Call 0 state=DISCONNCTD
 11:42:43.353  pjsua_media.c  Media session for call 0 is destroyed
 11:42:44.353  pjsua_media.c  Closing sound device after idle for 1 seconds
 11:42:44.353  pjsua_media.c  Closing Microsoft Sound Mapper - Output sound playback device and Microsoft Sound Mapper - Input sound capture device
 11:42:58.290   pjsua_core.c  RX 954 bytes Request msg INVITE/cseq=102 (rdata0x1b68864) from UDP 192.168.1.106:5060:
INVITE sip:1111 at 192.168.1.100:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK69042457;rport

Max-Forwards: 70

From: "2222" <sip:2222 at 192.168.1.106>;tag=as37bba9bb

To: <sip:1111 at 192.168.1.100:5060>

Contact: <sip:2222 at 192.168.1.106>

Call-ID: 4a3bf73b6bd880733eeb2c8a6475dca2 at 192.168.1.106

CSeq: 102 INVITE

User-Agent: Asterisk PBX 1.6.0.9-samy-r27

Date: Wed, 19 Aug 2009 04:43:23 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 397



v=0

o=root 2008801089 2008801089 IN IP4 192.168.1.106

s=Asterisk PBX 1.6.0.9-samy-r27

c=IN IP4 192.168.1.106

b=CT:384

t=0 0

m=audio 10282 RTP/AVP 0 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

m=video 17420 RTP/AVP 34 99

a=rtpmap:34 H263/90000

a=rtpmap:99 H264/90000

a=sendrecv


--end msg--
 11:42:58.291  pjsua_media.c  Media index 0 selected for call 1
 11:42:58.291   pjsua_core.c  TX 295 bytes Response msg 100/INVITE/cseq=102 (tdta0x1bb3a80) to UDP 192.168.1.106:5060:
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.1.106:5060;rport=5060;received=192.168.1.106;branch=z9hG4bK69042457

Call-ID: 4a3bf73b6bd880733eeb2c8a6475dca2 at 192.168.1.106

From: "2222" <sip:2222 at 192.168.1.106>;tag=as37bba9bb

To: <sip:1111 at 192.168.1.100>

CSeq: 102 INVITE

Content-Length:  0




--end msg--
 11:42:58.292  strm0x1b7a15c  VAD temporarily disabled
 11:42:58.292  strm0x1b7a15c  Encoder stream started
 11:42:58.293  strm0x1b7a15c  Decoder stream started
 11:42:58.293  pjsua_media.c  Media updates, stream #0: PCMU (sendrecv)
 11:42:58.293  pjsua_media.c  Opening sound device PCM at 16000/1/20ms
 11:42:58.398    ec0x1bb5498  AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=200 ms, latency=100 ms
 11:42:58.399   conference.c  Port 1 (sip:2222 at 192.168.1.106) transmitting to port 0 (Microsoft Sound Mapper - Input)
 11:42:58.399   conference.c  Port 0 (Microsoft Sound Mapper - Input) transmitting to port 1 (sip:2222 at 192.168.1.106)
 11:42:58.399   pjsua_core.c  TX 862 bytes Response msg 200/INVITE/cseq=102 (tdta0x1bb3a80) to UDP 192.168.1.106:5060:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.106:5060;rport=5060;received=192.168.1.106;branch=z9hG4bK69042457

Call-ID: 4a3bf73b6bd880733eeb2c8a6475dca2 at 192.168.1.106

From: "2222" <sip:2222 at 192.168.1.106>;tag=as37bba9bb

To: <sip:1111 at 192.168.1.100>;tag=ad4e2b7224ed4585a3ba37680f8f19f6

CSeq: 102 INVITE

Contact: <sip:1111 at 192.168.1.100:5060>

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

Supported: replaces, 100rel, norefersub

Content-Type: application/sdp

Content-Length:   327



v=0

o=- 3459670978 3459670979 IN IP4 192.168.1.100

s=pjmedia

c=IN IP4 192.168.1.100

t=0 0


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